/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
H A D | RTCEnumConverter.mm | 30 #include "talk/app/webrtc/peerconnectioninterface.h" 35 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState { 37 case webrtc::PeerConnectionInterface::kIceConnectionNew: 39 case webrtc::PeerConnectionInterface::kIceConnectionChecking: 41 case webrtc::PeerConnectionInterface::kIceConnectionConnected: 43 case webrtc::PeerConnectionInterface::kIceConnectionCompleted: 45 case webrtc::PeerConnectionInterface::kIceConnectionFailed: 47 case webrtc::PeerConnectionInterface::kIceConnectionDisconnected: 49 case webrtc::PeerConnectionInterface::kIceConnectionClosed: 55 (webrtc [all...] |
H A D | RTCSessionDescription+Internal.h | 30 #include "talk/app/webrtc/jsep.h" 31 #include "talk/app/webrtc/webrtcsession.h" 36 - (webrtc::SessionDescriptionInterface *)sessionDescription; 39 (const webrtc::SessionDescriptionInterface*)sessionDescription;
|
H A D | RTCICECandidate+Internal.h | 30 #include "talk/app/webrtc/peerconnectioninterface.h" 35 webrtc::IceCandidateInterface* candidate; 37 - (id)initWithCandidate:(const webrtc::IceCandidateInterface*)candidate;
|
H A D | RTCVideoRenderer+Internal.h | 30 #include "talk/app/webrtc/mediastreaminterface.h" 36 webrtc::VideoRendererInterface *videoRenderer; 38 - (id)initWithVideoRenderer:(webrtc::VideoRendererInterface *)videoRenderer;
|
H A D | RTCMediaConstraints.mm | 42 talk_base::scoped_ptr<webrtc::RTCMediaConstraintsNative> _constraints; 43 webrtc::MediaConstraintsInterface::Constraints _mandatory; 44 webrtc::MediaConstraintsInterface::Constraints _optional; 53 new webrtc::RTCMediaConstraintsNative(_mandatory, _optional)); 58 + (webrtc::MediaConstraintsInterface::Constraints) 60 webrtc::MediaConstraintsInterface::Constraints constraints; 62 constraints.push_back(webrtc::MediaConstraintsInterface::Constraint( 72 - (const webrtc::RTCMediaConstraintsNative *)constraints {
|
H A D | RTCEnumConverter.h | 32 #include "talk/app/webrtc/peerconnectioninterface.h" 37 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState; 40 (webrtc::PeerConnectionInterface::IceGatheringState)nativeState; 43 (webrtc::PeerConnectionInterface::SignalingState)nativeState; 46 (webrtc::MediaSourceInterface::SourceState)nativeState; 48 + (webrtc::MediaStreamTrackInterface::TrackState)convertTrackStateToNative: 52 (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
|
H A D | RTCPeerConnection+Internal.h | 33 #include "talk/app/webrtc/peerconnectioninterface.h" 38 talk_base::scoped_refptr<webrtc::PeerConnectionInterface> peerConnection; 41 talk_base::scoped_refptr<webrtc::PeerConnectionInterface>)peerConnection 42 observer:(webrtc::RTCPeerConnectionObserver *)observer;
|
/external/chromium_org/remoting/host/ |
H A D | screen_resolution.h | 10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 18 ScreenResolution(const webrtc::DesktopSize& dimensions, 19 const webrtc::DesktopVector& dpi); 22 webrtc::DesktopSize ScaleDimensionsToDpi( 23 const webrtc::DesktopVector& new_dpi) const; 26 const webrtc::DesktopSize& dimensions() const { return dimensions_; } 29 const webrtc::DesktopVector& dpi() const { return dpi_; } 36 webrtc::DesktopSize dimensions_; 37 webrtc::DesktopVector dpi_;
|
H A D | screen_resolution_unittest.cc | 16 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10)); 20 webrtc::DesktopSize(), webrtc::DesktopVector(10, 10)); 24 webrtc::DesktopSize(1, 1), webrtc::DesktopVector(0, 0)); 30 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10)); 32 EXPECT_TRUE(webrtc::DesktopSize(50, 50).equals( 33 resolution.ScaleDimensionsToDpi(webrtc [all...] |
H A D | chromoting_param_traits.cc | 12 void ParamTraits<webrtc::DesktopVector>::Write(Message* m, 13 const webrtc::DesktopVector& p) { 19 bool ParamTraits<webrtc::DesktopVector>::Read(const Message* m, 21 webrtc::DesktopVector* r) { 25 *r = webrtc::DesktopVector(x, y); 30 void ParamTraits<webrtc::DesktopVector>::Log(const webrtc::DesktopVector& p, 32 l->append(base::StringPrintf("webrtc::DesktopVector(%d, %d)", 37 void ParamTraits<webrtc::DesktopSize>::Write(Message* m, 38 const webrtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvoe.h | 33 #include "talk/media/webrtc/webrtccommon.h" 35 #include "webrtc/common_types.h" 36 #include "webrtc/modules/audio_device/include/audio_device.h" 37 #include "webrtc/voice_engine/include/voe_audio_processing.h" 38 #include "webrtc/voice_engine/include/voe_base.h" 39 #include "webrtc/voice_engine/include/voe_codec.h" 40 #include "webrtc/voice_engine/include/voe_dtmf.h" 41 #include "webrtc/voice_engine/include/voe_errors.h" 42 #include "webrtc/voice_engine/include/voe_external_media.h" 43 #include "webrtc/voice_engin [all...] |
H A D | webrtcvideodecoderfactory.h | 32 #include "webrtc/common_types.h" 34 namespace webrtc { namespace 44 virtual webrtc::VideoDecoder* CreateVideoDecoder( 45 webrtc::VideoCodecType type) = 0; 48 virtual void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) = 0;
|
H A D | webrtcvie.h | 33 #include "talk/media/webrtc/webrtccommon.h" 34 #include "webrtc/common_types.h" 35 #include "webrtc/modules/interface/module_common_types.h" 36 #include "webrtc/modules/video_capture/include/video_capture.h" 37 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" 38 #include "webrtc/modules/video_render/include/video_render.h" 39 #include "webrtc/video_engine/include/vie_base.h" 40 #include "webrtc/video_engine/include/vie_capture.h" 41 #include "webrtc/video_engine/include/vie_codec.h" 42 #include "webrtc/video_engin [all...] |
H A D | webrtcvideocapturer.h | 36 #include "talk/media/webrtc/webrtcvideoframe.h" 37 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" 38 #include "webrtc/modules/video_capture/include/video_capture.h" 47 virtual webrtc::VideoCaptureModule* Create( 49 virtual webrtc::VideoCaptureModule::DeviceInfo* CreateDeviceInfo(int id) = 0; 51 webrtc::VideoCaptureModule::DeviceInfo* info) = 0; 56 public webrtc::VideoCaptureDataCallback { 63 bool Init(webrtc::VideoCaptureModule* module); 80 webrtc::I420VideoFrame& frame); 82 webrtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | fakemediastreamsignaling.h | 32 #include "talk/app/webrtc/audiotrack.h" 33 #include "talk/app/webrtc/mediastreamsignaling.h" 34 #include "talk/app/webrtc/videotrack.h" 44 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling, 45 public webrtc::MediaStreamSignalingObserver { 48 webrtc::MediaStreamSignaling(talk_base::Thread::Current(), this) { 88 virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) { 90 virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) { 92 virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) { 94 virtual void OnAddLocalAudioTrack(webrtc [all...] |
H A D | testsdpstrings.h | 33 namespace webrtc { namespace 80 "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n" 142 } // namespace webrtc
|
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/ |
H A D | AudioSource.java | 28 package org.webrtc;
|
H A D | AudioTrack.java | 28 package org.webrtc;
|
H A D | StatsObserver.java | 28 package org.webrtc; 30 /** Interface for observing Stats reports (see webrtc::StatsObservers). */
|
H A D | VideoSource.java | 29 package org.webrtc;
|
/external/webrtc/src/system_wrappers/interface/ |
H A D | sleep.h | 15 namespace webrtc { namespace 22 } // namespace webrtc
|
/external/webrtc/src/system_wrappers/source/ |
H A D | cpu_no_op.cc | 15 namespace webrtc { namespace 22 } // namespace webrtc
|
H A D | critical_section.cc | 18 namespace webrtc { namespace 27 } // namespace webrtc
|
H A D | trace_unittest.cc | 17 using webrtc::CpuMeasurementHarness; 18 using webrtc::Trace; 19 using webrtc::kTraceWarning; 20 using webrtc::kTraceUtility; 22 class Logger : public webrtc::CpuTarget { 26 std::string trace_file = webrtc::test::OutputPath() + 29 Trace::SetLevelFilter(webrtc::kTraceAll);
|
/external/chromium_org/content/renderer/media/ |
H A D | mock_peer_connection_impl.h | 14 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" 21 class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { 26 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> 28 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> 31 webrtc::MediaStreamInterface* local_stream, 32 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; 34 webrtc::MediaStreamInterface* local_stream) OVERRIDE; 35 virtual talk_base::scoped_refptr<webrtc::DtmfSenderInterface> 36 CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE; 37 virtual talk_base::scoped_refptr<webrtc [all...] |